A Voice-over Internet Protocol (VoIP) system is established on the basis of both a Public Switched Telephone Network (PSTN) 11 and an Internet Protocol-Private Branch Exchange (IP-PBX) 1 that routes voice calls over the Internet.
A call entering the IP-PBX 1 from an Internet Protocol (IP) terminal 10 or a PSTN terminal 20 is connected to a destination IP terminal 10-1 through the following routes.
A separate signaling protocol or Session Initiation Protocol (SIP) is used in order to identify whether or not a call session of the IP terminal 10 is initiated/terminated.
When a SIP message as a call request message from the IP terminal 10 arrives at the IP-PBX 1, it is transferred to a SIP server program of the IP-PBX 1.
The SIP message is then transferred to the SIP client program, and the destination IP terminal 10-1 sends a SIP response message, such that a call is connected between the IP terminal 10 and the destination IP terminal 10-1.
Then, the IP terminal 10 and the destination IP terminal 10-1 exchange voices in the form of Real Time Protocol (RTP) packets through routes {circle around (c)} and {circle around (d)}.
Afterwards, when the call is terminated, a SIP message is exchanged and then the routes {circle around (a)} and {circle around (c)} are disconnected.
In call processing between the PSTN terminal 20 and the destination IP terminal 10-1, an off-hook message from the PSTN terminal 20 arrives at the IP-PBX 1 and is then transferred to the SIP server program of the IP-PBX 1.
In subsequence, the SIP message is transferred to the SIP client program, and the destination IP terminal 10-1 sends a SIP response message, so that a call is connected through routes {circle around (b)} and {circle around (c)}.
When the call is connected between the PSTN terminal 20 and the destination IP terminal 10-1 as above, voices from the PSTN terminal 20 are converted into data through a media Gateway Module (G/M) of the IP-PBX 1. The data is exchanged in the form of an RTP packet through the route {circle around (d)}, between the G/M of the IP-PBX 1 and the destination IP terminal 10-1.
Afterwards, when the call is terminated, a SIP message is exchanged and the call connection through the routes {circle around (c)} and {circle around (d)} is terminated.
A typical method of mirroring an RTP packet in the VoIP system including the IP-PBX of the related art includes preparing a mirroring apparatus for mirroring the RTP packet to a mirroring port of the IP-PBX 1, detecting, at the IP-PBX 1, whether or not a call has started up or terminated based on a SIP signal, transmitting a control signal based on the detection to the mirroring apparatus, and starting up or terminating, at the RTP packet, a mirroring procedure in response to the control signal.
However, when the IP-PBX does not have a function of providing the control signal, the mirroring apparatus is required to receive RTP packets transmitted/received through the IP-PBX 1 for all time periods, and store the RTP packets provided through the mirroring port over all time periods irrespective of session startup or termination.